System for configuration and status reporting of audio processing in TV sets

ABSTRACT

Systems are disclosed including a television (TV) set having included audio system. The TV set permits control over various functions, at least including audio volume, via a remote control. When the viewer activates the remote volume control, a graphic appears indicating the state of the volume control and, optionally the mute status. The graphic can be presented on the TV screen. In alternate embodiments, the graphic may be presented on a display of the remote, or even some other location, e.g., a different remote control. If a mute button is provided on the remote control, when the viewer activates the mute button, a graphic appears indicating the state of muting and, optionally, the volume control status. The TV set also offers control over various aspects of the audio system, including settings which go beyond volume up/down, generally through some sort of menu system.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a continuation-in-part of U.S. patent applicationSer. No. 15/862,045 entitled “Configurable Multi-Band CompressorArchitecture with Advanced Surround Processing,” filed Jan. 4, 2018,which claims priority to U.S. Provisional Patent Application No.62/442,195 entitled “Three Band Compressor with Advanced SurroundProcessing,” filed Jan. 4, 2017; this application also claims priorityto U.S. Provisional Patent Application 62/649,461 entitled “System forConfiguration and Status Reporting of Audio Processing in TV Sets,”filed Mar. 28, 2018; the entire content of each of which notedapplications is incorporated herein by reference.

BACKGROUND

It is common for television (TV) sets to offer remote control over awide range of parameters which determine details of the presentation toa TV viewer. These include universally common items such as channelselector, volume control, and audio mute, as well as more detailed andset-specific features as control over color, audio program selection,and automatic audio control features such as automatic loudness control,GEQ, surround etc.

It is nearly universal that when the remote volume control is activatedto turn volume up or down, or mute the audio, a visual indicator appearson the video display indicating the setting of the volume control or thepresence/absence of mute. In the former, this typically takes the formof a linear bar graphic, frequently accompanied by a number indicatingthe volume setting. Some manufacturers use a circular graphic with anumber instead. The latter typically takes the form of a representationof a speaker with or without an “X” superimposed over it.

In all these functions, when the viewer activates the remote control,the visual indicator appears while the remote is activated, and theindicator persists for a short time (usually a few seconds) after thelast activity in the remote. These indicators are useful for informingthe viewer of the current state of the volume and mute system in the TV.

In many TV sets, however, there are customarily additional featureswhich are accessible solely via a series of remote control activations.For example, the screen brightness is usually accessible by selecting a“menu” button, then using up/down arrows to select “video” from a listof options, pressing the “menu” button again (or an “enter” button),navigating via arrows to “brightness” and then using arrows to selectbrighter or darker. The setting is often displayed by means of a bargraph and number, as described above for the volume setting. However,these settings usually require the viewer to navigate through a complexmenu hierarchy, pressing more than one remote-control button, andsometimes the same button more than once, in order to reach the actualcontrol which changes the brightness. This “deep dive” into the menusystem results in these options typically being configured only once ornot at all.

Similarly, in modern TV sets, there are customarily additional audiofeatures which are accessible solely via series of remote-controlactivations. Many modern TV sets feature various sound “modes” such as“theatre,” “news,” “music,” and as well may allow activation of audiofeatures such as automatic loudness compensation, surround soundenhancement, and different frequency equalization options. These modesare accessible only through a similar deep dive into the remote-controlmenu system, wherein the viewer typically must press many buttons on theremote control, usually beginning with “menu” and leading on toselecting the audio submenu option, accessing the appropriate section ofthe audio submenu, and turning on/off or selecting the specific desiredaudio processing option. Often, if the wrong menu path is chosen, theviewer typically must either start over or navigate back up through themenu hierarchy.

It may be possible to add a button to the remote control which wouldprovide a shortcut to the audio sub-menu, thereby avoiding the need topress several buttons in sequence. Remote controls are, however,notorious for being cluttered with many buttons, and there are manyoptions which compete with each other for being controlled by a specificbutton. As well, additional buttons adds cost to the remote control. Onetrend in remote control system designs is to minimize the number ofbuttons presented to the viewer.

It may also be noted that discerning the state of the video display in aTV set is relatively easy within a very short stretch of time observinga TV picture. Low or excessive brightness, poor color settings, andimproper contrast are generally obvious upon short inspection. And,furthermore, many viewers find it unnecessary to change the detailedvisual settings that are accessible only through a deep dive into themenu system from program to program.

However, many more sophisticated audio settings are generally not aseasy to discern within a short period of time. For example, the presenceor absence of surround enhancement processing depends on the programmaterial to manifest itself. So, in order for a viewer to determinewhether surround enhancement processing is on or not requires eitherwaiting for the right program material, or conducting a time-intensiveand multi-step process (“taking a deep dive”) into the audio sub-menusystem to verify the processor's status. As well, since many of the moresophisticated audio settings would ideally be changed based on theprogram being viewed, this system requires the viewer to spend time onthe deep dive each time the program type is changed.

Requiring such a deep dive can be disadvantageous since it may result ina viewer being unaware of audio processing options available to improveenjoyment of listening, as well as potentially leaving the audio systemin a mode that is sub-optimum for the program being viewed.

SUMMARY

Embodiments of the technology described herein relate to adjusting soundcharacteristics in sound-producing devices and systems having multiplespeaker and two or more sound (auditory) channels, including but notlimited to television (TV) sets. Embodiments of the present disclosureprovide a number of advantages, relative to prior techniques, includingthe ability, when operating a television (TV) set, to avoid requiringmultiple button presses in order to get to the audio sub-menu within aTV set remote control. Embodiments of the present disclosure can make itpossible to avoid such multiple button presses without requiring aseparate button on the remote control associated with the audiosub-menu. Embodiments of the present disclosure can function to remindviewers of the status of audio options normally controlled by the audiosub-menu. Embodiments of the present disclosure can also function toinform viewers of available audio options which may improve theirviewing/listening experience.

One aspect of the present disclosure presents a system including atelevision (TV) set having included audio system. The TV set permitscontrol over various functions, at least including audio volume, via aremote control. When the viewer activates the remote volume control, agraphic appears indicating the state of the volume control and,optionally the mute status. If a mute button is provided on the remotecontrol, when the viewer activates the mute button, a graphic appearsindicating the state of muting and, optionally, the volume controlstatus. The TV set also offers control over various aspects of the audiosystem, including settings which go beyond adjusting sound volume(volume up/down), generally through some sort of menu system.

In exemplary embodiments, each time a remote volume control is pressed,the graphic which appears includes, in addition to the bar-graph,circular display, numerical readout, or other form of indicating volumelevel, another graphic that presents to the viewer the state ofadditional audio controls. In the preferred embodiment, these controlswill present the state of all first-level functions in the audio system.In the preferred embodiment, these controls allow selection of thesefirst-level features by simple navigation using up-down or left-rightarrows, then selecting menu or enter (or similar “activate” button) toturn them on or off or select the desired feature.

These, as well as other components, steps, features, objects, benefits,and advantages, will now become clear from a review of the followingdetailed description of illustrative embodiments, the accompanyingdrawings, and the claims.

BRIEF DESCRIPTION OF DRAWINGS

The drawings are of illustrative embodiments. They do not illustrate allembodiments. Other embodiments may be used in addition or instead.Details that may be apparent or unnecessary may be omitted to save spaceor for more effective illustration. Some embodiments may be practicedwith additional components or steps and/or without all of the componentsor steps that are illustrated. When the same numeral appears indifferent drawings, it refers to the same or like components or steps.

FIG. 1 shows a TV screen for a situation in which a viewer is watching arepresentative TV program.

FIG. 2 shows a viewer adjust the TV volume of the TV screen by using aremote control device.

FIG. 3 shows a visual indicator which appears when the volume control isactivated, e.g., pressed either up or down. In the case shown, Standardmode is the current TV audio sound mode configuration.

FIGS. 4A-4C show the visual indicator for a situation in which theviewer is changing the setting of the automatic volume control optionfor Standard mode.

FIGS. 5A-5C show the visual indicator for a situation in which theviewer is changing the setting of the pseudo-surround audio effect forStandard mode.

FIGS. 6A-6B show the display for a situation in which the viewerrestoring factory settings.

FIGS. 7A-7H show the visual indicator for situations in which the TV isconfigured in a sound mode other than Standard.

FIG. 8 is a box diagram showing components of an audio signal processingarchitecture and processing sequence according to an exemplaryembodiment of the present disclosure.

FIG. 9 depicts a block diagram of an exemplary embodiment of a dualprocessing protection (DPP) architecture in accordance with the presentdisclosure.

FIG. 10 depicts a diagram of an exemplary embodiment of a singlecompressor in accordance with the present disclosure.

FIG. 11A depicts a diagram of an exemplary embodiment of an advancedsurround (AS) architecture in accordance with the present disclosure.

FIG. 11B depicts a diagram of an exemplary delay-loop in accordance withthe present disclosure.

FIG. 12 depicts examples of delay-loop configurations according to anembodiment of the present disclosure.

FIG. 13 depicts a diagram of an exemplary embodiment of a staticequalizer (EQ) in accordance with the present disclosure.

FIG. 14 shows an example of the Multi Band Compressor Architecture ofFIG. 8 configured in bass-emphasized music mode.

DETAILED DESCRIPTION OF ILLUSTRATIVE EMBODIMENTS

Illustrative embodiments are now described. Other embodiments may beused in addition or instead. Details that may be apparent or unnecessarymay be omitted to save space or for a more effective presentation. Someembodiments may be practiced with additional components or steps and/orwithout all of the components or steps that are described and/or withthe component order changed.

Embodiments of the technology described herein provide a number ofadvantages, relative to prior techniques, including the ability, whenoperating a television (TV) set, to avoid requiring multiple buttonpresses in order to get to the audio sub-menu within a TV set remotecontrol. Embodiments of the present disclosure can make it possible toavoid such multiple button presses without requiring a separate buttonon the remote control associated with the audio sub-menu. Embodiments ofthe present disclosure can function to remind viewers of the status ofaudio options normally controlled by the audio sub-menu. Embodiments ofthe present disclosure can also function to inform viewers of availableaudio options which may improve their viewing/listening experience.

An aspect of the present disclosure provides a system including atelevision (TV) set having included audio system. The TV set permitscontrol over various functions, at least including audio volume, via aremote control. When the viewer activates the remote volume control, agraphic appears indicating the state of the volume control and,optionally the mute status. The graphic can be presented on the TVscreen. In alternate embodiments, the graphic may be presented on adisplay of the remote, or even some other location, e.g., a differentremote control. If a mute button is provided on the remote control, whenthe viewer activates the mute button, a graphic appears indicating thestate of muting and, optionally, the volume control status. The TV setalso offers control over various aspects of the audio system, includingsettings which go beyond volume up/down, generally through some sort ofmenu system.

In exemplary embodiments, each time the remote volume control ispressed, the graphic which appears includes, in addition to thebar-graph, circular display, numerical readout, or other form ofindicating volume level, another graphic that presents to the viewer thestate of additional audio controls. In the preferred embodiment, thesecontrols will present the state of all first-level functions in theaudio system. In the preferred embodiment, these controls allowselection of these first-level features by simple navigation usingup-down or left-right arrows, then selecting menu or enter (or similar“activate” button) to turn them on or off or select the desired feature.

FIG. 1 shows a TV screen 100 for a situation in which a viewer iswatching a representative TV program.

FIG. 2 shows the viewer adjusting the TV volume of the TV screen 100 byusing a remote control device 200.

FIG. 3 shows an exemplary visual indicator 300 according to the presentdisclosure, which preferably appears when a volume control, e.g., on theremote control device, is activated, e.g., pressed either up or down. Asshown, the state of the current sound mode (Standard) can be shown alongwith the state of the various audio effect components that make up thesound mode (e.g., Total Sonics, Total Volume, Total Surround). A briefdescription of the sound mode may be provided for the user (e.g.,viewer) and the user may be presented with the option to change thesound mode. Additionally, the user has the option to return to factorysettings. All of this information is available by a single action of theuser/view such as by activating, engaging, or selecting a controlmechanism or other activation means, e.g., a single selection with asingle push of the volume control button or other mechanism. In someembodiment, Standard mode, e.g., may optimize overall sound quality byincreasing bass, making dialog clear and natural, and widening thesoundfield.

FIGS. 4A-4C show the visual indicator 300 for an exemplary situation inwhich the viewer is changing the setting of the automatic volume controloption for Standard mode. In such a situation the viewer does not changethe sound mode but is able to change the characteristics of the soundlistening mode. In some embodiments the changes will be saved so thatthey can be recalled each time the modified sound mode is selected.

FIGS. 5A-5C show the visual indicator 300 for an exemplary situation inwhich the viewer is changing the setting of the pseudo-surround audioeffect for Standard mode. In these cases the viewer does not change thesound mode but is able to change the characteristics of the soundlistening mode. In some embodiments the changes will be saved so thatthey can be recalled each time the modified sound mode is selected.

FIGS. 6A-6B show the display 300 for an exemplary situation in which theviewer is restoring original or “factory” settings, such as when theuser decides to “undo” previous changes he or she has made and restorefactory settings.

FIGS. 7A-7H show the visual indicator 300 for exemplary situations inwhich the viewer adjusts volume and the TV is configured in a sound modeother than Standard; the figures also show means to change the soundmode, e.g., by highlighting the desired mode and pressing enter.

FIG. 7A depicts a Movie mode selection. Movie mode provides a surroundexperience that wraps around the listener, preserves clear dialog, andincreases low frequency response to hear rumbling sound effects. Itutilizes the L−R delay loop and width adjustment of or provided byadvanced surround to add a dimension of depth to the soundfield.

FIG. 7B depicts a Sports mode selection. Sports mode gives theannouncer's voice a prominent place in the mix while generating alifelike stadium atmosphere and crowd sound. It ensures the rightbalance between dialog and crowd/action noise.

FIG. 7C depicts a Music mode selection. Music mode emphasizes bass andhigh-frequency sounds, and is more natural and “spacious”.

FIG. 7D depicts a Concert Hall mode selection. Concert Hall modesimulates concert hall acoustics using advanced surround technology.

FIG. 7E depicts a Night mode selection. Night mode exercises tightcontrol over loudness variation so the viewer/user can understand ordiscern whispering dialog without worry of loud sound effects.

FIG. 7F depicts a News mode selection. News mode emphasizes dialog overother sound elements and controls loudness variation withoutcompromising wanted dynamics.

FIG. 7G depicts a Bass mode selection. Bass mode maximizes low-frequencyoutput

FIG. 7H depicts a User mode selection. In exemplary embodiments, Usermode allows the viewer to customize the television's audio settings forpreferred sound characteristics.

Exemplary Embodiments utilizing certain preferred TV set soundprocessing systems or architectures are described below and in relationto FIGS. 8-14; these are provided merely for example, and other systems,architectures, and embodiments can be realized and practiced within thescope of the present disclosure.

FIG. 8 is a box diagram showing components of an audio signal processingarchitecture 800 and processing sequence according to an exemplaryembodiment of the present disclosure. Architecture 800 includes a dualprocessing protection (DPP) block 802, a first cross over network(indicated as Crossover Network1) 804, first and second compressors(indicated as Compressor1 and Compressor2) 806 and 808, a first summingunit 810, an Advanced Surround block 812, and an EQ 814. As shown,architecture 800 also includes a second crossover network (indicated asCrossover Network2) 816, a third compressor (Compressor3) 818, ahigh-pass filter (HPF) 820, second summing unit 822, Soft Clip unit 824,and volume control unit 826. Representative input 1 and output 2 of thearchitecture 800 are also indicated. It should be noted that while asingle channel is shown for simplicity, e.g., by blocks 804, 806, 808,810, 816, 818, 820, 822 and 824, all connections between blocks shouldbe regarded as stereo, with both Right and Left stereo channels. Blocks802, 812, 814 and 826 are stereo input/output blocks inclusive. When thevolume control is positioned as shown in FIG. 8, the volume controlsetting can be configured to feedback to the compressors (as shown bydashed line). Exemplary configurable parameters for each component aredescribed below (of course others are within the scope of the presentdisclosure): (1) DPP: (L−R/L+R) Ratio Threshold and Center Gain; (2)Crossover network1: Crossover Frequency, Crossover Order; (3)Compressor1: Target Level, Noise Gate, Attack Threshold, ReleaseThreshold, Maximum Compressor Gain, Above Threshold Compression Ratioand Below Threshold Compression Ratio; (4) Compressor2: Target Level,Noise Gate, Attack Threshold, Release Threshold, Maximum CompressorGain, Above Threshold Compression Ratio, Below Threshold CompressionRatio and Coupling (with Compressor 1) adjustment; (5) AdvancedSurround: Width, Sum Feedback Delay, Sum Feedback Delay Coefficient, SumDelay Gain, Difference Feedback Delay, Difference Feedback DelayCoefficient, Difference Delay Gain, Diff Channel EQ Parameters; and (6)EQ: Center Frequency, Q and Gain for each of seven EQ filters; (7) oneor more additional compressors may also be included in certainembodiments and can be adjusted similar to as described above forCompressor1 and/or Compressor₂.

In exemplary embodiments, DPP 802, Crossover Network 1 804, Compressor1806 and Compressor2 808 can be configured as parts of a dynamic volumecontrol (DVC) system (or, architecture); such a DVC system can alsoinclude the EQ 814, Crossover Network2 816, Compressor3 818, and HPF 820configured for compressor-based bass enhancement. Examples of suitableEQs used with a crossover network and a compressor for dynamic volumecontrol include, but are not limited to, those disclosed in co-ownedU.S. Pat. No. 9,380,385 filed 14 Mar. 2014 and entitled “CompressorBased Dynamic Bass Enhancement with EQ,” the entire content of which isincorporated herein by reference. Examples of suitable DPPs used withcrossover networks and compressors for dynamic volume control include,but are not limited to, those disclosed in co-owned U.S. Pat. No.8,315,411 filed 16 Nov. 2009 and entitled “Dynamic Volume Control andMulti-Spatial Processing Protection,” the entire content of which isincorporated herein by reference. Another configuration, described inthe present disclosure, uses Advanced to Surround for a concert halleffect. While still another configuration utilizes the DPP TargetSum/Difference ratio, DPP center gain and Advanced Surround to create asports listening mode effect. And, still another configuration usesCompressor2 and Compressor3 together to create an improved bassenhancement effect.

FIG. 9 depicts a block diagram of an exemplary embodiment of a dualprocessing protection (DPP) architecture 900 in accordance with thepresent disclosure. Dual processing protection (DPP) is one form orembodiment of a more general Multi-Spatial Processing Protection (MPP),which can refer to the processing of two or more sound channels (e.g., Land R channels). Television manufacturers often include virtual surround(pseudosurround) technology (e.g., SRS Tru-Surround, Spatializer, etc.,or the like) in the two-channel television audio output path. Thistwo-channel television audio may go to speakers external to thetelevision or to speakers mounted in the television enclosure. Thesevirtual-surround technologies create the illusion of surround sound bymanipulating and enhancing the difference channel (L−R) present instereo broadcasts. The listener still perceives an intact center image(L+R) but also often hears the difference channel (L−R) either widenedover a broad soundstage or as a point source located somewhere otherthan the speaker locations. Often this type of spatial enhancement isdone during the production of the audio programming. This is especiallytrue of television commercials which are enhanced to grab the listener'sattention.

When an audio program has two cascaded stages of spatial enhancement(for example at the point of production and in a television's audioprocessing) there can be significant degradation in the audio quality.The preprocessed audio tends to have significant L−R energy relative toL+R energy. The second, cascaded stage, of spatial enhancementprocessing tends to increase the amount of L−R energy even more. Recentstudies have shown that excessive enhancement of L−R energy is one ofthe top factors in listener fatigue. There also can be a significantvolume increase. Accordingly, in accordance with one aspect of thepresent disclosure, a MPP system is provided. In exemplary embodiments,the MPP is a double processing protection (DPP) system that is a part ofa television audio signal reception and playback system, prior to thetelevision's stereo enhancement technology. The MPP system may bereferred to as a pseudosurround signal processor. An exemplary DPPsystem processes the audio signals so as to minimize the difference(L−R) enhancement (minimizing or reducing the energy level of thedifference (L−R) signal relative to the sum (L+R) signal) introduced atthe point of production. This allows the television's spatialenhancement technology to process the audio signals in a manner that ispsychoacoustically pleasing to the listener. The cascade of the DPPsystem before the television's spatial enhancement audio processing canbe quite effective in mitigating the harsh effects of double spatialprocessing. In some embodiments, the DPP system can be entirely digital,and/or can be implemented economically in software (e.g., C, C#,assembly language, etc.) and/or digital hardware (e.g., HDLdescription), etc. It should be appreciated that the DPP system can alsobe all analog, or a hybrid of analog and digital components.

The DPP 900 functions to limit the difference to sum ratio (L−R)/(L+R)based upon the Threshold setting. It should be noted that by adjustingthe Center gain, the sound field collapses proportionally into thecenter image, while boosting the sum channel, drawing the listener'sattention to the center image which is typically the program dialogue. Adetailed description of this function is provided in co-owned U.S. Pat.No. 8,315,411 filed 16 Nov. 2009 and entitled “Dynamic Volume Controland Multi-Spatial Processing Protection,” the entire content of which isincorporated herein by reference.

Referring to the DPP system 900 shown in FIG. 9, a left signal (L) andright signal (R) are respectively applied to the inputs 902 and 904 ofsystem 900. The L and R signals are applied to matrices represented bythe two signal summers 906 and 908. Signal summers 906 and 908constitute the matrix which provides the SUM (L+R) and DIF (L−R)signals. In the sum (L+R) path, the signal is generally untouched. TheSUM signal usually contains audio content which does not necessarilyneed to be localized. However, in alternate embodiments, frequencycontour shaping can be performed to enhance audio content such asdialogue. As shown, the SUM signal is multiplied by a Center constant atsignal multiplier 910 prior to be provided to matrices illustrated assignal summers 912 and 914. The Center constant allows the level of thecenter image (L+R) to be adjusted, if desired, to aid in intelligibilityof dialogue. Adding the L+R and L−R signals provides the left outputsignal Lo at output 916, while subtracting the L−R from the L+R providesthe right output signal Ro at output 918.

In the illustrated embodiment of FIG. 9, most of the processing occursin the DIF (L−R) path. L+R and L−R are compared to determine the levelof the L−R signal relative to L+R. Before comparison, these two SUM andDIF signals can be each passed through a respective high pass filter 920and 922, such as in circumstances where the speaker frequency responsedoes not include low frequencies. The L−R DIF signal can further bepassed through a multi-band equalizer 924 to accentuate the frequenciesthe human ear is most sensitive to, namely mid-range frequencies, tocompensate for the perceived loudness level of the L−R signal. Equalizer924 allows the difference channel level detection to be frequencydependent. For example, low frequency signals may be minimized whenprocessing for inexpensive television speakers with limited bassresponse. High frequencies may be minimized to limit the response totransient audio events. Typically mid-range frequencies, where the earis most sensitive, are equalized to dominate the difference leveldetection. Once the levels of the difference and sum signals arecalculated the DIF/SUM ratio is determined.

Each of these signals is then run through a respective signal leveldetector 928 and 930. The detectors listed above can be used, such as anRMS level detector, although any type of level detector (such as theones described above) can be used. Also, the processing can all beperformed in the log domain to increase efficiency by processing themthrough the log domain processing blocks 932-1 and 932-2.

The outputs of the blocks 932-1 and 932-2 are applied to the signalsummer wherein the processed SUM signal is subtracted from the processedDIF signal. Subtracting one signal from the other in the log domain isthe same as providing a signal that is the ratio of the process SUMsignal to that of the DIF signal in the linear domain. Once the L+R andL−R signal levels are calculated, where the L−R signal level may havebeen equalized prior to level detection to increase the mid-rangefrequencies, these two signal levels are compared by the comparator 938to a preset threshold 940. The ratio between the two signals((L−R)/(L+R)) is compared to a threshold ratio by comparator 938 inorder to determine the recommended L−R signal gain adjustment. A limiterstage 942 may be used to limit the amount and direction of gain appliedto the L−R signal. The illustrated embodiment limits the gain at 0 dBhence only allowing attenuation of the L−R signal, although in someapplications, there may be a desire to amplify the L−R signal. Anaveraging stage 944 averages, with a relatively long time constant, theoutput of the limiter stage 942 so as to prevent the DPP system fromtracking brief transient audio events. After conversion back to thelinear domain by linear domain block 946, the level of the L−R signal iscorrespondingly adjusted by the signal multiplier 948 to achieve thattarget ratio.

FIG. 10 depicts a diagram of an exemplary embodiment of a compressorarchitecture (or, system) 1000, e.g., as can be used for Compressor1,Compressor2 and/or Compressor3 for the embodiment of FIG. 8 (DVC system800). A detailed description of similar compressor architecture isprovided in co-owned U.S. Pat. No. 8,315,411. As shown in FIG. 10,architecture 1000 receives two input signals, a left signal L at input1002 and a right signal at input 1004. In exemplary embodiments, the DVCsystem architecture may be based upon a digital implementation of aclassic compressor design (THAT Corporation Design Note 118) withflexibility and additional modifications that are only possible in adigital implementation. Architecture or system 1000 may also be referredto a DVC architecture or system.

Architecture 1000 can include an RMS level detector 1010 for providing asignal representative of the sum of the RMS averages of the left andright signals L and R, log conversion block 1012, and a signal averagingAVG block 1014. Log conversion block 1012 converts the output of the RMSlevel detector 1010 from the linear to the logarithmic domain. System1000 is responsive to a number of control signals each indicative ofwhether a certain condition exists requiring a response from the system.The system 1000 may also include a host processor (not shown) configuredand arranged for carrying out the operating of the DVC system 1000. Theillustrated embodiment is responsive to a number of control signalsincluding: a target level signal provided by the target signalgenerating device 1016, an attack threshold signal generated by theattack threshold signal device 1018, a release threshold (not shown), agate threshold signal generated by the gate threshold signal device1020, an attack ratio threshold (not shown), a release ratio threshold(not shown), a ratio signal generated by the ratio signal device 1022,and a mute hold signal generated by mute hold device 1024 responsive toa program change detector (PCD—not shown). Devices (or components) 1016,1018, 1020, 1022 can simply be adjustable user controls accessible tothe user. Device 1024 can be arranged to receive a signal from the TVcontrols when the channel changes or from a mute detector (not shown)that detects if inputs 1002 and 1004 have both been muted. The targetsignal level 1016 represents the level in dB, relative to a full scaleinput, that is the target volume. The attack threshold 1018 representsthe number of dB that REF (output of summer 1026) must be above AVGbefore the attack time is reduced by a factor of N, where N can be anynumber. In one illustrated embodiment N=10.

The release threshold signal preferably represents the number of dB thatREF must be below AVG before the release time is reduced by a factor ofM, where M can be any number, and in one illustrated embodiment M=10.The Gate threshold 1020 represents the amount (e.g., a negative dBnumber) that REF can go below AVG before all left and right gainadjustments are frozen. The attack ratio threshold represents theabsolute amount, in dB, that REF can go above the target signal level1016 before the volume control begins attenuating the input signal. Therelease ratio threshold represents the absolute amount, in dB, that REFcan go below the target signal level 1016 before the volume controlbegins adding gain to the input signal. The ratio signal 1022 adjuststhe AVG value by the desired compression ratio.

Target level signal 1016 is subtracted from the output of log conversionblock 1012 by signal summer 1026 so as to provide the REF signal to thesignal averaging AVG block 1014, a comparator 1028 and a secondcomparator 1030. The REF signal represents the volume level of the inputsignal relative to the desired listening threshold. The AVG signal canalso be thought of as the instantaneous (prior to attack/releaseprocessing) ideal gain recommendation. The output of the signalaveraging block 1014 is the AVG signal, which is a signal that is afunction of the average of the REF signal. The AVG signal is applied tothe signal summer 1032 where it is added to the attack threshold signal1018. In a similar manner (not shown) the AVG signal is summed with arelease threshold. The AVG signal is also applied to the signal summer1034 where it is added to the gate threshold signal 1020. The output ofsignal summer 1032 is applied to attack threshold comparator 1028 whereit is compared to the REF signal, while the output of signal summer 1034is applied to gate threshold comparator 1030 where it is compared to theREF signal. The AVG signal is also multiplied by the ratio signal 1022by the signal multiplier 1036. The output of comparator 1028 is appliedto the attack/release selection block 1038, which in turn provideseither an attack (ATT) signal, or a release (REL) signal to the signalaveraging block 1014, dependent on and responsive to the status of themute hold 10 signal 1024. The output of the release threshold AVG summer(not shown) is also compared to the REF signal and is applied to theattack/release selection block. The comparator 1030 provides an outputto the HOLD input of signal averaging block 1014. Finally, the signalmultiplier 1036 provides an output to a log-to-linear signal converter1040, which in turn provides an output which is applied to each of thesignal multipliers 1042 and 1044, wherein it respectively scales theleft and right signal provided at the corresponding inputs 1002 and 1004so as to provide the output modified left and right signals Lo and Ro.

With continued reference to FIG. 10, the RMS level detector 1010 sensesthe sound level of the input signal. It should be noted that while anRMS level detector is shown, any type of signal level detector can beused. For example, a peak detector, average detector, perception basedlevel detector (such as the ITU 1770 loudness detector or the CBSloudness detector), or other detector can be used to sense the soundlevel. These level detectors usually have time constants which aredynamically and independently adjustable. One method of adjusting thesetime constants is to base them on the envelope or general shape of theinput signal so that the time constants vary with the signal. In otherembodiments, the time constants are fixed. For ease of data processing,the sound level can be converted into the log domain, as shown, usinglog conversion block 1012. In a multi-band system, a separate RMSdetector can be used for each band. The signal averaging block 1014 isconfigured and arranged so as to compute the average of REF relative tothe attack and release times. The output signal AVG of the signalaveraging block 1014 is adjusted by the desired compression ratio, viamultiplier 1036, to create the gain value to be applied. Finally thegain is converted back into the linear domain by the log-to-linearconverter 1040 for application to the left and right signals L and R soas to produce the modified left and right signals Lo and Ro.

A target output level represented by the target level signal 1016 issubtracted from the sensed level at the output of the log conversionblock 1012 to determine the difference between the actual and desiredsound level. This difference, which represents the level of the inputsignal relative to the target level signal 1016, is known as thereference (REF) signal. The target level signal can be a user input,such as a simple knob or other pre-set setting, so as to control thelevel of sound desired. This threshold can be fixed or it can be changedas a function of the input signal level to better position thecompression relative to the input dynamic range. Once REF signal isobtained, it is provided as an input to the averaging block 1014, attackthreshold comparator 1028 and gate threshold comparator 1030. The outputof attack threshold comparator 1028 is applied to the attack/releaseselect block 1038, which in turn can receive a signal (e.g., a MuteHoldsignal 1024) from a program change detector.

The gate threshold signal 1020 when added to the current average AVGrepresents the lowest value REF is able to achieve before left and rightgain adjustment (1042 and 1044) are frozen. The gate thresholdcomparator 1030 receives the instantaneous signal level (REF) signal anddetermines if the sound level represented by REF drops below the givenaforementioned threshold. If the instantaneous signal level (REF) ismore than the amount of the gate threshold below the averaged signallevel (AVG) appearing at the output of block 1014, the gain applied tothe signal in the signal path is held constant until the signal levelrises above the threshold. The intent is to keep the system 1000 fromapplying increased gain to very low level input signals such as noise.In an infinite hold system, the gain can be constant forever until thesignal level rises. In a leaky hold system, the gain can be increased ata gradual pace (much slower than the release time). In one embodiment,this gate hold threshold is adjustable, while in another embodiment thethreshold set by gate threshold 1034 is fixed. A detailed description ofsimilar suitable compressor architecture is provided in co-owned U.S.Pat. No. 8,315,411, which is incorporated by reference herein in itsentirety.

The architecture 1000 preferably (but not necessarily) has an adjustablemaximum limit to the gain applied to the L and R channel. By limitingthe maximum gain, one can minimize the effects of compressor overshootwhen the source material transitions from very quiet to very loud suchas when a television program transitions to a loud commercial.Additionally, a maximum gain limit allows one to minimize the noiseboost that can occur when the audio is quiet. This is especiallyimportant for analog input sources or older program material that has ahigh noise floor.

In some embodiments, the DPP 802, Crossover Network1 804, Compressor1806 and Compressor2 808 components can be configured as a volume controlwith multi-spatial processing protection similar to as described in U.S.Pat. No. 8,315,411. Examples of suitable compressor blocks (or,architectures or subsystems) include, but are not limited to, thosedisclosed in co-owned U.S. Pat. No. 8,315,411.

The Volume Control 826 setting is provided to Compressor1 andCompressor2 (dashed line on FIG. 8) as an optional means toautomatically adjust the Compressor target level 1016 as a function ofthe system Volume Control setting. This Volume Control feedback would bevaluable if the compressors were configured as signal level limiterswith high compression (1000:1 ratio) above threshold and no compression(1:1 ratio) below threshold. As the volume level is decreased thefeedback would allow the compressor target level to increase thusensuring that the maximum allowed signal level is always possible at thespeaker terminals. Conversely, as the volume level increases thecompressor targets can be lowered to ensure that the maximum allowedsignal will not be exceeded at the speaker terminals.

FIG. 11A depicts a diagram of an exemplary embodiment of an AdvancedSurround architecture/system 1100. FIG. 11B shows a detailed diagram ofthe delay loop. Architecture 1100 includes left and right channels 1102and 1104; signal flows are indicated by arrows. As shown in FIG. 11A,the Advanced Surround architecture 1100 includes summers (summing units)1106 and 1108, each of which receives the Left and Right channels 1102and 1104. Summer 1106 is configured to invert one input so iteffectively acts as a subtracting unit. The summing units 1106 and 1108produce difference 1107 and sum 1109 channels as outputs, respectively.Difference EQ 1110 in the difference channel 1107 preferably focuses onmiddle frequencies, which are the ones the human ear is most sensitiveto; it may be desirable in some applications to single out or separateout those so they will dominate the spatialization, spreading and addingmultiple dimensions to the spatialization. The difference channel alsoincludes a delay loop 1112 and multipliers 1114 and 1116 for impressingwidth and gain inputs/values; summer 1118 is also present to combine thewidth adjusted difference channel with the diff delay gain adjusteddifference delay output. The sum channel 1109 can include a HPF filter1122 to filter out low frequency signals as those signals typically donot add much to perceived spatialization; HPF filter 1122 is preferablypresent but is optional. The sum channel 1109 can also include a delayloop 1124 and a summer 1126 for setting (impressing) a delay gain.Summing units 1120 and 1128 are also present.

Referring to FIG. 11B, an exemplary embodiment of a delay loop, e.g.,1112 and 1124 in FIG. 11A, is shown. The delay loop can include a summer1130, a delay unit 1132, and a multiplier 1134 for setting or impressinga feedback delay coefficient. Other architectures may be used for delayloops within the scope of the present disclosure. The Delay Looparchitecture is repeated as the L−R Delay Loop and L+R Delay Loop. TheAdvanced Surround parameters Width, Diff Delay Gain, Sum Delay Gain,Delay (in Delay Loop) and Feedback Delay Coefficient (in Delay Loop) allcorrespond to adjustable parameters. Setting the Diff Delay Gain and SumDelay Gain to zero (i.e., no signal passes) transforms the algorithm tobe similar to a pseudo-surround two channel processing algorithm asdisclosed in co-owned U.S. patent application Ser. No. 12/949,397, whichalgorithm is typically used in many consumer electronics applications.Co-owned U.S. patent application Ser. No. 12/949,397 filed 18 Nov. 2010and entitled “Virtual Surround Signal Processing” is incorporated in itsentirety herein by reference. With continued reference to FIGS. 11A and11B, the Width parameter adjusts the level of sound field spread as istraditionally done. The Delay Loop provides an example of a means tomodel near and far reflections of both the Sum and Diff signals.

FIG. 12 provides more detail on how one might configure the sum anddifference delay loops for exemplary embodiments. FIG. 12.1 shows how toconfigure the Delay, e.g., for 20 ms or 30 ms based upon a 48 kHz samplerate; other sampling rates may of course be utilized. FIG. 12.2 showsthe impact of the Delay and feedback coefficient settings on the DelayLoop impulse response. The Delay setting determines the time gap betweenthe non-zero values of the impulse response. For stability, the absolutevalues of the Feedback Delay Coefficients are preferably limited tovalues greater than or equal to 0 and less than 1; other coefficientsmay be used, however. The higher the absolute value of the FeedbackDelay Coefficient, the slower the decrease of the impulse response. FIG.12.3 defines the early reflection and reverb areas of an acousticimpulse response. One can see that by appropriate choice of the Delay,Feedback Delay Coefficient and the Delay Gain the impulse response ofthe L+R and L−R can be independently configured to position energy, asneeded, in the early reflection and late reflection (reverberant)regions.

This architecture allows the summation of a scaled amount of sum anddifference reflection/reverb with processing before the entire signal iscombined back with the left and right channels. Prior art algorithms maybe adequate for spreading the perceived sound field, for two stereospeakers, in the horizontal direction. The addition of reflection/reverbmodelling, as shown in FIGS. 11A and 11B, adds depth, some height andadditional width perception to the virtual sound field resulting in aperceived 3D sound effect. The trade-off in providing thiswidth/depth/height expression, via digital delay reflection and reverbmodelling, is audio clarity. It preferably is used with less emphasis(subtly) when the desire is to provide more fidelity to inexpensivespeakers; and, it preferably is used with emphasis (more strongly) whenthe desire is to create the ambience of a concert hall, theater orsporting event. It should be noted that more sophisticatedreflection/reverb modelling techniques can be used, such as those thatmodel the impulse responses of theaters and concert halls, to produce aneven more configurable and pleasing effect (though with much increasedimplementation complexity).

FIG. 13 depicts a diagram of an exemplary embodiment of the Static EQ1300. Left and right channels 1310 and 1320 are indicated as havingconfigurable parametric EQ 1312 and 1322, respectively. In theembodiment shown, seven configurable parametric EQ second-order sectionsare configured in both left and right channels. Of course, otherembodiments of a static EQ, e.g., of different order (e.g., 3^(rd),4^(th), 5^(th), 6^(th), 7^(th), 8^(th), etc.) and number of sections(e.g., 1, 2, 3, 4, 5, 6, etc.) may be utilized within the scope of thepresent disclosure; moreover, dynamic EQ may be used in addition orsubstitution.

One preferred embodiment of Compressor3 involves Volume Controlfeedback. The Volume Control setting can be provided, as feedback, toCompressor3 818 (dashed line on FIG. 8) as an optional means toautomatically adjust the Compressor target level (Level 1016 in FIG. 10)as a function of the system Volume Control setting. For example, thisvolume level feedback is useful when Compressor3 818 is configured in abass enhancement configuration. As the volume level is decreased thefeedback would allow the compressor target level to increase thusensuring that the maximum bass level is always possible at the speakerterminals. Conversely, as the volume level increases the compressortargets can be lowered to ensure that the maximum allowed bass levelwill not be exceeded at the speaker terminals. An embodiment utilizesCompressor2 808 in conjunction with Compressor3 818 to provide evenbetter system bass response in a bass-emphasized music configuration. Adiagram of another exemplary embodiment of Crossover Network2 816,Compressor3 818 and HPF 820, in a Bass Enhancement configuration, isdescribed in detail in co-owned U.S. Pat. No. 8,315,411, the entirecontent of which is incorporated herein by reference.

The preferred instantiation of the Soft Clipper is a hard limiterfollowed by a smoothing polynomial. Suitable smoothing polynomialsinclude, but are not limited to, the type described in the paperEsqueda, F., et al., 23rd European Signal Processing Conference,“Aliasing reduction in soft-clipping algorithms,” EUSIPCO 2015 (Dec. 22,2015): 2014-2018, a copy of which is submitted with and incorporatedinto this application; one such suitable the polynomial isy=(3×/2)(1−x²/3), where y is the clipper output, is utilized in apreferred static soft clipping instantiation. Other smoothingpolynomials and methods may be used, e.g., other methods based on theideal bandlimited ramp function (BLAMP), or the polyBLAMP polynomialapproximation method, etc. A hard clipper alone can produce a harshaudio artifact during compressor overshoot. A true limiter may becomputationally intensive and require significant processor bandwidthand memory. A soft clipper represents a good compromise that minimizesthe perceived audio artifacts for brief audio excursions above fullscale.

This configurable multi-compressor (e.g., three-compressor) system canbe utilized to enhance the listener experience for different types ofprogram material. For example, it can be configured in a music mode withan emphasis on bass. It can be configured in a concert hall mode withemphasis on echo and reverberation. It can also be configured in a livesporting event mode that emphasizes the announcer's voice whilemaintaining the ambience of a stadium environment. There are many otherpossible configurations such as HiFi, News and Theater modes, etc.

Embodiments of the present disclosure can accordingly improve thefidelity and perceived sound field spread of inexpensive, cabinetmounted, stereo speakers such as those that might be found intelevisions, wireless speaker systems and sound bars. Embodiments of thepresent disclosure can improve inexpensive, cabinet mounted, stereospeakers by providing, e.g., (i) an Advanced Surround algorithm thatadds depth and height to the left/right/center sound field images, (ii)a Soft Clip algorithm to minimize the perceived artifacts caused bycompressor overshoot, (iii) configurable crossover filter orderadjustment to allow better isolation between bands, (iv) a compressormaximum gain adjustment to reduce overshoot and minimize noise boost,and/or (v) a center gain adjustment to emphasize the perception of thecenter image (dialog) in high ambient sound situations.

It may be desirable to have different configurations of or settings(adjustments) of these architectures and/or algorithms, depending uponthe type of audio equipment or source material. For example, whilewatching an action movie, a listener may be interested in a strong audiosurround effect. Embodiments of the present disclosure can accordinglyprovide enhanced audio surround effect(s). As another example, whenlistening to music, a listener may be less interested in a surroundeffect and more interested in high fidelity, a concert hall effect, orincreased bass. A listener to a sporting event may be interested inhearing the announcer clearly over the crowd noise and public addresssystem while still trying to maintain the ambiance of a stadiumenvironment. The improvements and configurability of the architecturesand algorithms of the present disclosure can thus provide theimplementation of multiple audio enhancement modes to facilitatedifferent types of audio material and the listener's taste.

Listening Modes: exemplary embodiments including exemplary listeningmodes are described below with respect to features and componentsdescribed above and shown in the drawings; other listening modes may ofcourse be realized in accordance with the present disclosure.

Music Mode:

An example of a bass-emphasized music mode will now be described. Anexemplary music mode can be created by the following configurations andsettings; others may be used within the scope of the present disclosure.It is assumed that the system utilizes an inexpensive set of speakerswhich have a low-end frequency response that extends to, e.g., about 250Hz. This mode can utilize two compressors (Compressor2 and Compressor3)and EQ to enhance bass.

FIG. 14 shows an example of the Multi Band Compressor Architecture 800of FIG. 8 configured in a bass-emphasized music mode (“MUSIC Mode”). Anexample bass-emphasized music mode configuration is described below.

For the MUSIC Mode, the DPP 802 may be configured to limit the(L−R)/(L+R) ratio to 0 dB. Compressor1 806 may be configured separatelyto limit the level in the mid and high bands. In this example those aresignals above, e.g., 250 Hz. The high band (>250 Hz) Above ThresholdRatio (compression ratio) is set to 1000:1 to provide true limiting atthe Target Level. The Target Level 1016 (FIG. 10) is determined, whilemonitoring the speaker output, with the EQ 814 configured and with theTV Volume Control 826 set at full volume to determine the maximumallowable signal. The Target Level 1016 will increase proportionally,via internal feedback, as the TV volume is decreased. In other words,the high band compressor will allow more energy to pass as the volumecontrol is lowered since it will be attenuated by the volume controlprior to being present at the speaker terminals. The Max Gain and BelowThreshold Ratio setting (e.g., 1.2:1) will allow some mid and high bandboost to occur when the input level, in conjunction with the TV volumecontrol 826 setting, indicates more energy will be tolerated. In otherwords the high band compressor will allow more mid and high frequencyenergy as the volume control is lowered since it will be attenuated bythe volume control prior to being present at the speaker terminals.

Compressor2 808 may be configured to limit (or boost) the level in thelow band relative to a target level setting. In this case the low bandcould be 250 Hz and below. Crossover Network1 804 is configured at 250Hz. The filter order is set to 4^(th) to optimize the separation of thetwo bands (<250 Hz and >250 Hz). The Target Level 1016 (FIG. 10) setsthe limit, for this band, in dB full scale. The Target Level 1016 isset, while monitoring the speaker output, with the Volume Control 826 atfull volume and with the EQ 814 configured with any desired static boost<250 Hz. Setting the Target Level 1016 in this manner allows the maximumamount of energy <250 Hz allowable (before distortion occurs) to reachthe speaker terminals at full volume. At lower volume settings theVolume Control feedback will allow more bass signal to pass. Thisconfiguration allows the system to always pass as much bass signal aspossible, without distortion, while utilizing EQ to provide a staticboost to the low band. The Maximum Compressor Gain could be set to a lowvalue (e.g., 2-3 dB) to allow a small amount of additional dynamic boostat low bass input levels. Above and below threshold compression ratiosare set relatively high (e.g., 16:1).

The Advanced Surround 812 may be configured with a moderate to smallamount of sound field spread (Width) with a Delay, Delay Feedback andDelay Gain configuration that is dominated by early reflections givingthe subtle feeling of 3D sound without sacrificing clarity. The EQ 814may be configured to flatten the speaker frequency response in mid tohigh bands and to boost the response in the low band. This creates goodoverall tonal balance while providing the desired amount of bass boost.

Compressor3 118 may be configured to limit very low frequency signals(<<250) that are not passable by the speakers at high, or even moderate,output levels. This lower low band is set in Crossover Network2 804, forexample, it is set to 100 Hz. The Target Level can be set to a level(lower than the Compressor2 808 Target Level 1016) that will allow thesevery low frequency signals to still pass (at limited levels) and evenboost them, via the Max Gain and Below Threshold Ratio parameters, ifthe input signal level and volume control 826 setting will allow. TheTarget Level is set when the TV is at full volume, while monitoring thespeaker output, to determine the maximum allowable signal but will thenbe increased proportionally, via internal feedback, as the TV volume isdecreased. In other words the low-low band compressor will allow moreenergy to pass as the volume control is lowered since it will beattenuated by the volume control prior to being present at the speakerterminals. The HPF is preferably configured to remove those extremelylow frequencies that absolutely cannot be reproduced by the speaker.Soft Clip 824 may be configured to limit signals above 0 dB full scale.By dividing the speaker low band into two bands, the configurationdescribed above allows lower than typical frequencies to be passed bythe speakers. In prior art, a HPF would typically be used to remove thelower-low band frequencies from the audio signal. This new compressorconfiguration allows them to be passed if conditions (low input level,low volume control setting) merit. All these parameter settings arecalibrated for a given set of speakers mounted in a specific enclosure.

Concert Hall Mode:

An exemplary concert hall mode can be created, for the examplespeaker(s), by the following configurations and settings; others may beused within the scope of the present disclosure.

DPP 802: Same as bass-emphasized music mode. Compressor1 806: Same asbass-emphasized music mode. Compressor2 808: Same as bass-emphasizedmusic mode. EQ 814: Same as bass-emphasized music mode. Compressor3 818:Same as bass-emphasized music mode. Soft Clip 824: Same asbass-emphasized music mode.

Advanced Surround 812: Increase the Delay Time and Delay FeedbackCoefficient for both the L+R and L−R channels so that the overallimpulse response extends well into the reverberation region.

Broadcast Sports Mode (Sports Mode):

An exemplary broadcast sports mode can be created by the followingconfigurations and settings; others may be used within the scope of thepresent disclosure. In exemplary embodiments, the above-describedarchitecture(s), e.g., as depicted for FIGS. 8-14, can be configuredspecifically in a Sports Mode tuning for broadcast events that occur instadiums, arenas, etc., where there is dialogue (e.g., announcers,commentators, etc.) and typically a considerable amount of unpredictableambient noise (e.g., cheering crowds). The Sports Mode can operate toreduce the amount of ambient audio, to bring attention to the dialogue,but then processing the vestige of difference signal that remains, afterDPP processing, with a reverb effect, to create a subtle perception of astadium.

DPP 802: Limit the (L−R)/(L+R) ratio to, e.g., −12 dB. This settingreduces the ambient audio (such as crowd noise, public addressannouncer, etc.). Increase the Center Gain (via multiplier 916) toemphasize the broadcast announcer's voice. This gives the announcer'svoice more perceived clarity without sacrificing the overall bandwidthof the audio signal. As a result the difference signal is attenuated inhigh ambient-noise environments (e.g., stadiums, arena etc.). Incontrast, prior art techniques/systems have simply implemented abandpass filter to pass voice frequencies while attenuating signalsoutside of the voice range; but, without additional processing thesporting event ambience can be lost, resulting in a loss of the livesporting event ambience and/or less clarity in the broadcast announcer'svoice, which is something that has been observed with the prior art.Crossover Network1 804, Compressor1 806 and Compressor2 808 can beconfigured in a dynamic volume control (DVC) configuration.

Advanced Surround 812: Configure L−R Delay loop (Delay and DelayFeedback Coefficient) to generate an impulse in the reverberationregion. Disable L+R delay loop 1124 by setting the sum delay gain to,e.g., 0. While the L−R channel is reduced by the DPP—forming a reducedremaining difference signal, the reverberation on the reduced remainingdifference signal restores the enveloping feel of stadium crowd noisewithout sacrificing the clarity of the broadcast announcer. Disablingthe L+R delay loop 1124 can help maintain the broadcast announcer'svocal clarity. EQ 814: Configure to compensate for speaker frequencyresponse and to provide bass boost.

Compressor3 818: Configured to improve the speaker's bass response bylimiting (or boosting) the level in the low band relative to a targetlevel setting. The Target Level 1016 is set with the Volume Control atfull volume and with the EQ fully configured with the desired low badboost. Setting the Target Level in this manner allows the maximum amountof energy (before distortion occurs) to reach the speaker terminals atfull volume. At lower volume settings the Volume Control feedback willallow more bass signal to pass. This configuration allows the system toalways pass as much bass signal as possible, without distortion, whileutilizing EQ to boost the low band. The HPF is configured to remove thelow frequencies that cannot be reproduced by the speaker in thisconfiguration. Soft Clip 824: Configured to limit signals above, e.g., 0dB full scale.

In exemplary embodiments (not just applicable to a broadcast sportsmode), another included feature is volume control feedback (VCF) todynamically adjust the target levels for the three compressors. This maybe needed or desirable when the Volume control is positioned after acompressor that is configured to function as a limiter. The TV VolumeControl signal or setting can be provided, as feedback, to a compressoras an optional means to automatically adjust the Compressor target levelas a function of the system Volume Control. For example, this volumelevel feedback is useful when Compressor3 818 is configured in a bassenhancement configuration. As the volume level is decreased the feedbackwould allow the compressor target level to increase thus ensuring thatthe maximum bass level is always possible at the speaker terminals.Conversely, as the volume level increases the compressor targets can belowered to ensure that the maximum allowed bass level will not beexceeded at the speaker terminals.

Exemplary Clauses

Clause 1: A system for enhancing stereo audio adjustment in TV sets, thesystem including: a sound processing system configured to provide audiocontrols to modify the characteristics of the processed audio; a displayfor observation by a user and configured to display volume controlinformation and sound settings corresponding to user-selectable soundprocessing characteristics; and a control device configured to adjustthe sound processing characteristics in response to user input; and,activation means to activate the display via a single selection on thecontrol device.

Clause 2: The system of clause 1, wherein the system is configurable toproduce a desired sound listening mode.

Clause 3: The system of clause 1, wherein the system is configurable tochange the characteristics of a given sound listening mode.

Clause 4: The system of clause 2, further including: a dual processingprotection (DPP) processor configured to limit the audio difference tosum ratio (L−R)/(L+R) by (i) attenuating the audio difference signal,producing a reduced remaining difference signal, and (ii) adjusting thelevel of the sum signal; and wherein the desired listening mode is anaudio processing configuration operative to enhance a listener'sperception of a center image in stereo audio, while maintaining aperception of a surrounding difference channel.

Clause 5: The system of clause 4, further including a two channelsurround processor configured to set the amount of perceived sound fieldspread in multiple dimensions by filtering the reduced remainingdifference signal from the DPP processor and generating perceived audioreflections of the filtered signal.

Clause 6: The system of clause 2, wherein the sound processing systemincludes: an equalizer configured to shape the input audio frequencyresponse for the desired spectral characteristics; a crossover networkto divide the input audio into two bands, the first band being the lowerband and the second band being the upper band; a first compressor forprocessing the first band configured to produce output signalsresponsive to the energy level of the lower band frequency portion ofthe input audio signal; a second compressor for processing the secondband configured to produce output signals responsive to the energy levelof the upper band frequency portion of the input audio signal; a thirdcompressor with a configurable target level determined dynamically bysystem volume control feedback; a high pass filter configured to removefrequencies that cannot be reproduced by the loudspeaker; a summer torecombine first and second bands; and a soft clipper operative to limitthe perceived distortion of left and right audio signals that brieflyexceed a full scale output.

Clause 7: The system of clause 2, wherein the sound processing systemincludes at least one compressor with a configurable target leveldetermined dynamically by system volume control feedback; and a volumecontrol positioned after the compressor.

Clause 8: The systems of any of the foregoing clauses in anycombination.

Accordingly, from the above-given description and accompanying drawings,it will be apparent that this technology described herein can allow,among other things, a user (viewer) to adjust the sound mode and soundconfiguration, on a program-by-program basis, without taking a deep diveinto the on-screen menu hierarchy. This technology described herein(disclosed invention) gives the typical viewer more control of the TVaudio configuration and makes the viewing experience much morepleasurable.

The components, steps, features, objects, benefits, and advantages thathave been discussed are merely illustrative. None of them or the relateddiscussions are intended to limit the scope of protection in any way.Numerous other embodiments are also contemplated. These includeembodiments that have fewer, additional, and/or different components,steps, features, objects, benefits, and/or advantages. These alsoinclude embodiments in which the components and/or steps are arrangedand/or ordered differently.

For example, in bass-emphasized MUSIC Mode the roles of Compressor2 andCompressor3 could be reversed. Compressor2 could compress the lower lowband and Compressor3 could compress the upper low band. Additionally,the HPF could be located after the summer. The Volume Control could bepositioned before Crossover Network2 eliminating the need for VolumeControl feedback.

Unless otherwise stated, all measurements, values, ratings, positions,magnitudes, sizes, and other specifications, including frequencies,ratios, and dB values, that are set forth in this specification,including in the claims that follow, are approximate and/or provided asexample, are not necessarily exact or invariable. They (the valuesdescribed) are intended to have a reasonable range that is consistentwith the functions to which they relate and with what is customary inthe art to which they pertain.

All articles, patents, patent applications, and other publications thathave been cited in this disclosure are incorporated herein by reference.

The phrase “means for” when used in a claim is intended to and should beinterpreted to embrace the corresponding structures and materials thathave been described and their equivalents. Similarly, the phrase “stepfor” when used in a claim is intended to and should be interpreted toembrace the corresponding acts that have been described and theirequivalents. The absence of these phrases from a claim means that theclaim is not intended to and should not be interpreted to be limited tothese corresponding structures, materials, or acts, or to theirequivalents.

The scope of protection is limited solely by the claims that now follow.That scope is intended and should be interpreted to be as broad as isconsistent with the ordinary meaning of the language that is used in theclaims when interpreted in light of this specification and theprosecution history that follows, except where specific meanings havebeen set forth, and to encompass all structural and functionalequivalents.

Relational terms such as “first” and “second” and the like may be usedsolely to distinguish one entity or action from another, withoutnecessarily requiring or implying any actual relationship or orderbetween them. The terms “comprises,” “comprising,” and any othervariation thereof when used in connection with a list of elements in thespecification or claims are intended to indicate that the list is notexclusive and that other elements may be included. Similarly, an elementproceeded by an “a” or an “an” does not, without further constraints,preclude the existence of additional elements of the identical type.

None of the claims are intended to embrace subject matter that fails tosatisfy the requirement of Sections 101, 102, or 103 of the Patent Act,nor should they be interpreted in such a way. Any unintended coverage ofsuch subject matter is hereby disclaimed. Except as just stated in thisparagraph, nothing that has been stated or illustrated is intended orshould be interpreted to cause a dedication of any component, step,feature, object, benefit, advantage, or equivalent to the public,regardless of whether it is or is not recited in the claims.

The abstract is provided to help the reader quickly ascertain the natureof the technical disclosure. It is submitted with the understanding thatit will not be used to interpret or limit the scope or meaning of theclaims. In addition, various features in the foregoing detaileddescription are grouped together in various embodiments to streamlinethe disclosure. This method of disclosure should not be interpreted asrequiring claimed embodiments to require more features than areexpressly recited in each claim. Rather, as the following claimsreflect, inventive subject matter lies in less than all features of asingle disclosed embodiment. Thus, the following claims are herebyincorporated into the detailed description, with each claim standing onits own as separately claimed subject matter.

What is claimed is:
 1. A system for enhancing stereo audio adjustment ina TV set having two or more speakers, the system comprising: a volumecontrol configured to adjust or mute a volume of a TV set, wherein thevolume control is further configured to adjust sound processingcharacteristics in response to user input; a dual processing protectionprocessor (DPP) operative to receive left (L) and right (R) signals fromleft (L) and right (R) audio channels, respectively, as inputs andconfigured to limit the audio difference to sum ratio (L−R)/(L+R) byattenuating the audio difference signal (L−R), thereby producing anattenuated difference signal, and increasing the level of the sum signal(L+R), wherein the level of the sum signal (L+R) represents center gaincorresponding to a center image relative to the two or more speakers,wherein the dual processing protection processor is operative to produceleft and right output signals for the TV set; at least one crossovernetwork configured to receive left and right input signals and toseparate each of the left and right input signals into a plurality offrequency bands, including a high band and a low band, wherein the atleast one crossover network produces left and right output signals foreach band and wherein the at least one crossover network is configurableby a user to set a crossover frequency and filter order of the crossovernetwork; first and second compressors, wherein the first compressor isconfigured to receive the left and right high band output signals fromthe first crossover network and to produce compressed left and righthigh band output signals corresponding to the high band, and wherein thesecond compressor is configured to receive the left and right low bandoutput signals from the first crossover network and to producecompressed left and right low band output signals corresponding to thelow band, and wherein the first and second compressors are configured tolimit a level of the high and low bands; a two-channel surroundprocessor configurable to receive the outputs from the first and secondcompressors and set an amount of perceived sound field spread inmultiple spatial dimensions based on the attenuated difference signal(L−R) resulting from processing of the dual processing protectionprocessor and to generate left and right output signals; a stereoequalizer configured to receive left and right audio input signals andshape the spectral characteristics such that the overall bandwidth ofthe audio material is not compromised when subjected to the playback onthe two or more speakers, wherein the stereo equalizer is configured toproduce left and right output signals; a soft clipper operative toreceive as inputs compressed stereo left and right input signals,wherein the soft clipper is configured to limit the perceived distortionof left and right signals that briefly exceed a full scale output whensubjected to the center gain, and dynamics of the first and secondcompressors and to produce a stereo output; a display for observation bya user and configured to display volume control information of thevolume control and sound settings corresponding to user-selectable soundprocessing characteristics; and activation means to activate the displayvia a single selection on the control device.
 2. The system of claim 1,wherein the system is configurable to produce desired sound listeningmodes.
 3. The system of claim 1, wherein the filter order of the atleast one crossover network is 4th order.
 4. The system of claim 1,wherein the system further comprises a third compressor configured tolimit the amount of low frequency audio energy in a band defined by alow frequency response of a speaker of the system, and with a limitthreshold determined dynamically by a volume control setting of thesystem, and configured to dynamically boost the audio level when theaudio level is below the limit threshold, wherein the system isconfigured for bass-enhanced music listening.
 5. The system of claim 1,wherein the volume control is configured to receive a stereo input andto feed the volume control setting forward to at least one of the stereocompressors.
 6. The system of claim 1, wherein the volume control isconfigured to receive a stereo input and to feed the volume controlsetting back to at least one of the stereo compressors.